path: root/include/oboe/AudioStreamBuilder.h

/*
 * Copyright 2015 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#ifndef OBOE_STREAM_BUILDER_H_
#define OBOE_STREAM_BUILDER_H_

#include "oboe/Definitions.h"
#include "oboe/AudioStreamBase.h"

namespace oboe {

/**
 * Factory class for an audio Stream.
 */
class AudioStreamBuilder : public AudioStreamBase {
public:

    AudioStreamBuilder() : AudioStreamBase() {}

    AudioStreamBuilder(const AudioStreamBase &audioStreamBase): AudioStreamBase(audioStreamBase) {}

    /**
     * Request a specific number of channels.
     *
     * Default is kUnspecified. If the value is unspecified then
     * the application should query for the actual value after the stream is opened.
     */
    AudioStreamBuilder *setChannelCount(int channelCount) {
        mChannelCount = channelCount;
        return this;
    }

    /**
     * Request the direction for a stream. The default is Direction::Output.
     *
     * @param direction Direction::Output or Direction::Input
     */
    AudioStreamBuilder *setDirection(Direction direction) {
        mDirection = direction;
        return this;
    }

    /**
     * Request a specific sample rate in Hz.
     *
     * Default is kUnspecified. If the value is unspecified then
     * the application should query for the actual value after the stream is opened.
     *
     * Technically, this should be called the "frame rate" or "frames per second",
     * because it refers to the number of complete frames transferred per second.
     * But it is traditionally called "sample rate". Se we use that term.
     *
     */
    AudioStreamBuilder *setSampleRate(int32_t sampleRate) {
        mSampleRate = sampleRate;
        return this;
    }

    /**
     * Request a specific number of frames for the data callback.
     *
     * Default is kUnspecified. If the value is unspecified then
     * the actual number may vary from callback to callback.
     *
     * If an application can handle a varying number of frames then we recommend
     * leaving this unspecified. This allow the underlying API to optimize
     * the callbacks. But if your application is, for example, doing FFTs or other block
     * oriented operations, then call this function to get the sizes you need.
     *
     * @param framesPerCallback
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setFramesPerCallback(int framesPerCallback) {
        mFramesPerCallback = framesPerCallback;
        return this;
    }

    /**
     * Request a sample data format, for example Format::Float.
     *
     * Default is Format::Unspecified. If the value is unspecified then
     * the application should query for the actual value after the stream is opened.
     */
    AudioStreamBuilder *setFormat(AudioFormat format) {
        mFormat = format;
        return this;
    }

    /**
     * Set the requested buffer capacity in frames.
     * BufferCapacityInFrames is the maximum possible BufferSizeInFrames.
     *
     * The final stream capacity may differ. For AAudio it should be at least this big.
     * For OpenSL ES, it could be smaller.
     *
     * Default is kUnspecified.
     *
     * @param bufferCapacityInFrames the desired buffer capacity in frames or kUnspecified
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setBufferCapacityInFrames(int32_t bufferCapacityInFrames) {
        mBufferCapacityInFrames = bufferCapacityInFrames;
        return this;
    }

    /**
     * Get the audio API which will be requested when opening the stream. No guarantees that this is
     * the API which will actually be used. Query the stream itself to find out the API which is
     * being used.
     *
     * If you do not specify the API, then AAudio will be used if isAAudioRecommended()
     * returns true. Otherwise OpenSL ES will be used.
     *
     * @return the requested audio API
     */
    AudioApi getAudioApi() const { return mAudioApi; }

    /**
     * If you leave this unspecified then Oboe will choose the best API
     * for the device and SDK version at runtime.
     *
     * If the caller requests AAudio and it is supported then AAudio will be used.
     *
     * @param audioApi Must be AudioApi::Unspecified, AudioApi::OpenSLES or AudioApi::AAudio.
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setAudioApi(AudioApi audioApi) {
        mAudioApi = audioApi;
        return this;
    }

    /**
     * Is the AAudio API supported on this device?
     *
     * AAudio was introduced in the Oreo 8.0 release.
     *
     * @return true if supported
     */
    static bool isAAudioSupported();

    /**
     * Is the AAudio API recommended this device?
     *
     * AAudio may be supported but not recommended because of version specific issues.
     * AAudio is not recommended for Android 8.0 or earlier versions.
     *
     * @return true if recommended
     */
    static bool isAAudioRecommended();

    /**
     * Request a mode for sharing the device.
     * The requested sharing mode may not be available.
     * So the application should query for the actual mode after the stream is opened.
     *
     * @param sharingMode SharingMode::Shared or SharingMode::Exclusive
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setSharingMode(SharingMode sharingMode) {
        mSharingMode = sharingMode;
        return this;
    }

    /**
     * Request a performance level for the stream.
     * This will determine the latency, the power consumption, and the level of
     * protection from glitches.
     *
     * @param performanceMode for example, PerformanceMode::LowLatency
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setPerformanceMode(PerformanceMode performanceMode) {
        mPerformanceMode = performanceMode;
        return this;
    }


    /**
     * Set the intended use case for the stream.
     *
     * The system will use this information to optimize the behavior of the stream.
     * This could, for example, affect how volume and focus is handled for the stream.
     *
     * The default, if you do not call this function, is Usage::Media.
     *
     * Added in API level 28.
     *
     * @param usage the desired usage, eg. Usage::Game
     */
    AudioStreamBuilder *setUsage(Usage usage) {
        mUsage = usage;
        return this;
    }

    /**
     * Set the type of audio data that the stream will carry.
     *
     * The system will use this information to optimize the behavior of the stream.
     * This could, for example, affect whether a stream is paused when a notification occurs.
     *
     * The default, if you do not call this function, is ContentType::Music.
     *
     * Added in API level 28.
     *
     * @param contentType the type of audio data, eg. ContentType::Speech
     */
    AudioStreamBuilder *setContentType(ContentType contentType) {
        mContentType = contentType;
        return this;
    }

    /**
     * Set the input (capture) preset for the stream.
     *
     * The system will use this information to optimize the behavior of the stream.
     * This could, for example, affect which microphones are used and how the
     * recorded data is processed.
     *
     * The default, if you do not call this function, is InputPreset::VoiceRecognition.
     * That is because VoiceRecognition is the preset with the lowest latency
     * on many platforms.
     *
     * Added in API level 28.
     *
     * @param inputPreset the desired configuration for recording
     */
    AudioStreamBuilder *setInputPreset(InputPreset inputPreset) {
        mInputPreset = inputPreset;
        return this;
    }

    /** Set the requested session ID.
     *
     * The session ID can be used to associate a stream with effects processors.
     * The effects are controlled using the Android AudioEffect Java API.
     *
     * The default, if you do not call this function, is SessionId::None.
     *
     * If set to SessionId::Allocate then a session ID will be allocated
     * when the stream is opened.
     *
     * The allocated session ID can be obtained by calling AudioStream::getSessionId()
     * and then used with this function when opening another stream.
     * This allows effects to be shared between streams.
     *
     * Session IDs from Oboe can be used the Android Java APIs and vice versa.
     * So a session ID from an Oboe stream can be passed to Java
     * and effects applied using the Java AudioEffect API.
     *
     * Allocated session IDs will always be positive and nonzero.
     *
     * Added in API level 28.
     *
     * @param sessionId an allocated sessionID or SessionId::Allocate
     */
    AudioStreamBuilder *setSessionId(SessionId sessionId) {
        mSessionId = sessionId;
        return this;
    }

    /**
     * Request an audio device identified device using an ID.
     * On Android, for example, the ID could be obtained from the Java AudioManager.
     *
     * By default, the primary device will be used.
     *
     * Note that when using OpenSL ES, this will be ignored and the created
     * stream will have deviceId kUnspecified.
     *
     * @param deviceId device identifier or kUnspecified
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setDeviceId(int32_t deviceId) {
        mDeviceId = deviceId;
        return this;
    }

    /**
     * Specifies an object to handle data or error related callbacks from the underlying API.
     *
     * <strong>Important: See AudioStreamCallback for restrictions on what may be called
     * from the callback methods.</strong>
     *
     * When an error callback occurs, the associated stream will be stopped and closed in a separate thread.
     *
     * A note on why the streamCallback parameter is a raw pointer rather than a smart pointer:
     *
     * The caller should retain ownership of the object streamCallback points to. At first glance weak_ptr may seem like
     * a good candidate for streamCallback as this implies temporary ownership. However, a weak_ptr can only be created
     * from a shared_ptr. A shared_ptr incurs some performance overhead. The callback object is likely to be accessed
     * every few milliseconds when the stream requires new data so this overhead is something we want to avoid.
     *
     * This leaves a raw pointer as the logical type choice. The only caveat being that the caller must not destroy
     * the callback before the stream has been closed.
     *
     * @param streamCallback
     * @return pointer to the builder so calls can be chained
     */
    AudioStreamBuilder *setCallback(AudioStreamCallback *streamCallback) {
        mStreamCallback = streamCallback;
        return this;
    }

    /**
     * If true then Oboe might convert channel counts to achieve optimal results.
     * On some versions of Android for example, stereo streams could not use a FAST track.
     * So a mono stream might be used instead and duplicated to two channels.
     * On some devices, mono streams might be broken, so a stereo stream might be opened
     * and converted to mono.
     *
     * Default is true.
     */
    AudioStreamBuilder *setChannelConversionAllowed(bool allowed) {
        mChannelConversionAllowed = allowed;
        return this;
    }

    /**
     * If true then  Oboe might convert data formats to achieve optimal results.
     * On some versions of Android, for example, a float stream could not get a
     * low latency data path. So an I16 stream might be opened and converted to float.
     *
     * Default is true.
     */
    AudioStreamBuilder *setFormatConversionAllowed(bool allowed) {
        mFormatConversionAllowed = allowed;
        return this;
    }

    /**
     * If set to None then Oboe will not do sample rate conversion. But the underlying APIs
     * might do sample rate conversion. Unfortunately sample rate conversion in Android typically
     * prevents one from getting a low latency stream. So we can do the conversion in Android
     * and still get a low latency stream.
     *
     * Default is SampleRateConversionType::Sinc. TODO currently Linear
     */
    AudioStreamBuilder *setSampleRateConversionQuality(SampleRateConversionQuality quality) {
        mSampleRateConversionQuality = quality;
        return this;
    }

    /**
     * @return true if AAudio will be used based on the current settings.
     */
    bool willUseAAudio() const { // TODO How can this be const if isSupported() is not const?
        return (mAudioApi == AudioApi::AAudio && isAAudioSupported())
                || (mAudioApi == AudioApi::Unspecified && isAAudioRecommended());
    }

    /**
     * Create and open a stream object based on the current settings.
     *
     * The caller owns the pointer to the AudioStream object.
     *
     * @param stream pointer to a variable to receive the stream address
     * @return OBOE_OK if successful or a negative error code
     */
    Result openStream(AudioStream **stream);

protected:

private:

    /**
     * Create an AudioStream object. The AudioStream must be opened before use.
     *
     * The caller owns the pointer.
     *
     * @return pointer to an AudioStream object or nullptr.
     */
    oboe::AudioStream *build();

    AudioApi       mAudioApi = AudioApi::Unspecified;
};

} // namespace oboe

#endif /* OBOE_STREAM_BUILDER_H_ */